Hello folks,

By means of Asterisk 11 and sip.conf, I got success implementing early media. That is, all information that come from callee (SIP 183 message/ SDP) is passed to the caller without any modification in the SDP body. However, in Asterisk 13 and using pjsip.conf I’m still failing to do the same thing. See:

Softphojne1 < ------------------------------------------------> Asterisk < --------------------------------------->Softphone2

| ———————-SIP INVITE ————————>|

| —————————-SIP INVITE————>|

. . .

. .

. .

|< ----------SIP 183 ---------------------------------| SDP : Media Description, name and address (m): audio 4000 RTP/AVP 8 96 Media Description, name and address (m): video 5000 RTP/AVP 97 |< --------------------------SIP 183-----------------------------| SDP :Media Description, name and address (m): audio 13258 RTP/AVP 8 96 Media Description, name and address (m): video 16002 RTP/AVP 97 So, there is bit modification in SDP body, caused by Asterisk. As long as I’m intending to implement direct media, I believe that Asterisk 13 has some special configuration to be done in PJSIP.conf file, that will allow things work very well again, as in Asterisk 11 and sip.conf. How to configure pjsip.con file or Asterisk, to run direct-media? Or , where to find a tutorial about it on Internet? Any hint will be very helpful!